Re: [cisco-voip] on boarding new employees

[Pavan K]

Look at unimax second nature (2N).
It has service now integration and a pretty solid product.

On Wed, Apr 25, 2018, 12:22 PM Scott Voll  wrote:

> Does anyone have a good solution for on boarding new employees?
>
> we have to:
>
> create a 7961 EM profile
> create a 8861 EM profile
> Create a IP Communicator EM Profile
> create an IP communicator device
> associate the user EM profiles to the AD account
> create an Unity connection VM box
> set the zero out option.
>
> and if they are on the contact center there is some other stuff to.
>
> I know that through API's I could do something but I don't have that
> kind of time to create something. .
>
> What are options these days?
>
> UC products are all 11.x
>
> TIA
>
> Scott
>
>
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[cisco-voip] PSA: CCMP 11.6

[Pavan K]

For those brave enough to be running ccmp and looking to upgrade to 11.6,

Agent-id aka PeripheralID can no longer start with a 0.

There is now a hardcoded limit of 72k (configured) agents on the large
deployment model.
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[cisco-voip] CTL cert - migration to diff cluster

[Pavan K]

Folks,
Could use some help with a migration issue..

Ucm 8.6 encrypted with tokens that are no longer available.
Have about 2k 7945 phones with CTL and ITL installed.

Trying to migrate these phones to a diff nonsecure cluster running ucm 10.

I was able to get the ITL file erased using the rollback enterprise
parameter.

Every time I delete the CTL file, phone reboots and downloads it again.
Phone security profile has been updated to non secure and there is no LSC
on the phones.

Need the CTL file off the phone long enough to switch vlans.

What are my options ?
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Re: [cisco-voip] CAD vs Finesse Differences

[Pavan K]

It will take a couple of revisions for finesse to work like CAD. Till then
we are stuck with it.

I have a lot of customers who have elected to wait on 10.6 for finesse to
stabilize.

By the way I have nothing against finesse and  have several customers on
finesse, just not the big ones..

On Dec 2, 2016 3:01 PM, "JASON BURWELL" <jason.burw...@foundersfcu.com>
wrote:

> Thanks for the additional information Anthony! What I found interesting
> was the wording of the introduction in document. Is there really a decision
> to be made? I thought 10.6 was the end of the line for CAD?
>
>
>
> “By learning how these differences affect your business and
> agent/supervisor productivity, you will be able to make a well-informed
> decision about which solution is best for your business.”
>
>
>
>
>
> Jason
>
>
>
>
>
> *From:* avhollo...@gmail.com [mailto:avhollo...@gmail.com] *On Behalf Of 
> *Anthony
> Holloway
> *Sent:* Friday, December 02, 2016 3:02 PM
> *To:* Pavan K <pav.c...@gmail.com>
> *Cc:* JASON BURWELL <jason.burw...@foundersfcu.com>; Cisco VoIP Group <
> cisco-voip@puck.nether.net>
> *Subject:* Re: [cisco-voip] CAD vs Finesse Differences
>
>
>
> I was about to blast that document for being really old, but then I
> clicked it, and I now see that it was updated last month.  That's
> encouraging.  At least Spark hasn't stolen the UCCX documentation team away
> from us.
>
>
>
> The document is still missing a few key things though.
>
>
>
> As one example, CAD use to show the bread crumb trail for the call, so if
> you wanted to know which number they dialed, queue they arrived to you
> from, and if they spoke to any other agents, or were in any other CSQs,
> that was all in CAD by default.  Finesse doesn't have this.  So, you need
> to fake it with Enterprise Data Fields, and it's still not as good, plus
> you have to have some complex scripting using session variables, etc.  It's
> not really practical to expect us Engineers to just boiler plate that in to
> every Finesse greenfield.
>
>
>
> Then, there's probably other things you need to consider too, which are
> not exactly a comparison of clients, but rather a list of items which need
> to be addressed on Finesse.  For example, if you're a Supervisor and a
> Reporting person, your Finesse Live Data gadgets will now show every single
> Agent and CSQ, instead of just those for which you supervise.  You'll need
> to follow this reference
> <http://cp.mcafee.com/d/k-Kr3xEgdEIfn7nud7dTPtPqadPhOqemnAS7PqadPhOqemnCkrCQkkrLfCzAsepdI3zhOUeodyeQH3y3sj7DDwFSHs4fzhSPvbCTqJMg-d7rdYKrd7ara2b3P_nVyX2aoUtRXBQSjhOZMVxd6WpEVVqWtAklrTjVkffGhBrwqrhdI6XCXCM0lFkJnBYJDMNKCkrmmH6u8lJjbdQDlnUBXlffQxmePBm6dMAXbQ2ZGMDJFBXv2vNb-7Hh-BchhuujpbP-Ne5p-nrJofbTLMedWp9RffGoQI2y8DOVIQsI3AjobZ8Qg6BKQGmGncRAIqnjh0920sq83d3h0bOxZauq80WjHLN-5LEropdP0sJqybbzTu_>
> to have that corrected.
>
>
>
> On Fri, Dec 2, 2016 at 8:55 AM, Pavan K <pav.c...@gmail.com> wrote:
>
> http://www.cisco.com/c/en/us/products/collateral/customer-
> collaboration/finesse/white-paper-c11-730883.html
> <http://cp.mcafee.com/d/5fHCNAi3zqb3RNRTzhPtYTsSyzsQsCzBBVdxYSyzsQsCzBBVB6VJ556XPVEV73Cjqaaab0UQsK3C3ozJaMUwT4NVVUatGT13UQtITOVJSHs4fzhSPvbCPhOCOwyMY_R-oKMyCe7tuVtdAQsLseojhKCqeumKDp55mZQ-l3PWApmU6CQPr1KVKVI05qlblVvbpYcrFyQqEdSDjblffSvNQ_bdQDlnUBXlffQxmePBm6eOxZavxwDavBRJm-9uOa9ssPtVVDwsrQOjGuvkNFo54hfBPpEVo78CMnWhEwdbtFkJkKpH9oQKCy0i40UQg6q6y0nB3WkYQg1QDnvzYbvgSMOr1l9V>
>
>
>
>
>
> On Dec 2, 2016 8:47 AM, "JASON BURWELL" <jason.burw...@foundersfcu.com>
> wrote:
>
> Is there a summary or matrix that shows the feature differences between
> CAD and Finesse that would be on UCCX 10.6x? I have seen threads mentioning
> some feature deficiencies with earlier versions of Finesse but would like
> to know the overall differences on current versions.  Thanks, Jason
>
>
>
>
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>
>
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>
>
>
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Re: [cisco-voip] call-forward

[Pavan K]

I believe you can remove the callfwdall softkey using an ephone template
On May 4, 2016 6:40 PM, "Quenten Grasso"  wrote:

> Hi,
>
>
>
> I’m trying to work out if it’s possible to stop the user of a 7960 phone
> from using the call-forward feature one CCME.
>
>
>
> as they seem to keep setting a call-forward by accident and not realising.
>
>
>
> Regards,
>
> Quenten Grasso
>
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Re: [cisco-voip] Looking for beta testers for UCCX Call Flow Designer

[Pavan K]

Aah nice. Didn't realize we could generate visios from aef. Sign me up
please.

Thanks.
On May 3, 2016 10:56 PM, "Anthony Holloway" <avholloway+cisco-v...@gmail.com>
wrote:

> Value is hard to place properly, as it's so subjective.
>
> What makes this an even harder value proposition is that it's from a
> smaller third party company (sorry, no offense), and a customer who invests
> in this tool, will have to bet on it being around for the life of the
> system.  Else, it was simply just a good idea at the time.
>
> Where I see this really adding value is in green field deployments where
> you can involve BAs in the documentation process, all the while they are
> piecing together the building blocks for the actual script.  Or also in
> daily operations, where an update to the documentation, translates directly
> into the change itself.  This eliminates documentation of call flows
> becoming out of date.  And since you can use this tool to simply create
> images or Visio diagrams, without the need to export AEF files, you can
> technically use it for any and all call flow diagrams (I.e., Hunt Groups)
> keeping a consistent look and feel to your documents.
>
> I'd like to try this tool out, and I have an upcoming project that's not
> too large where I think this could be a good fit.  Reach out to me Tanner.
>
> On Tue, May 3, 2016 at 9:41 PM, Kevin Przybylowski <kev...@advancedtsg.com
> > wrote:
>
>> I can see value if it will take existing AEFs and export them to a
>> functional Visio diagram.
>>
>>
>>
>> *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On
>> Behalf Of *Pavan K
>> *Sent:* Tuesday, May 3, 2016 9:31 PM
>> *To:* Tanner Ezell <tanner.ez...@gmail.com>; Cisco VOIP <
>> cisco-voip@puck.nether.net>
>> *Subject:* Re: [cisco-voip] Looking for beta testers for UCCX Call Flow
>> Designer
>>
>>
>>
>> I watched the video and am curious. What is the value add to uccx editor ?
>>
>>
>>
>>
>>
>> On Tue, May 3, 2016, 8:24 PM Tanner Ezell <tanner.ez...@gmail.com> wrote:
>>
>> I now realize I forgot to include the video!
>> https://www.youtube.com/watch?v=CUwGGbPjmWY
>>
>>
>>
>> On Tue, May 3, 2016 at 3:34 PM, Tanner Ezell <tanner.ez...@gmail.com>
>> wrote:
>>
>> As many of you know I've been developing a software solution to bring
>> visual call flow development to UCCX (you can see a demo video I put
>> together showcasing functionality, including export to Visio and AEF).
>> After much time I'm extremely pleased to announce we're looking for beta
>> testers to play with the software and provide feedback.
>>
>>
>>
>> Ideally I'm looking for folks who are able to apply real world needs to
>> the application to make recommendations for improvement and enhancements.
>>
>>
>>
>> Anyone interested please feel free to email me directly or on thread to
>> discuss further.
>>
>>
>>
>> The software is Web based, so if you're a Mac or Linux guy, you can still
>> apply!
>>
>>
>>
>> I'm also happy to answer any questions out there regarding the software.
>>
>>
>>
>> Regards,
>>
>> Tanner Ezell
>>
>>
>>
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Re: [cisco-voip] Looking for beta testers for UCCX Call Flow Designer

[Pavan K]

I watched the video and am curious. What is the value add to uccx editor ?



On Tue, May 3, 2016, 8:24 PM Tanner Ezell  wrote:

> I now realize I forgot to include the video!
> https://www.youtube.com/watch?v=CUwGGbPjmWY
>
> On Tue, May 3, 2016 at 3:34 PM, Tanner Ezell 
> wrote:
>
>> As many of you know I've been developing a software solution to bring
>> visual call flow development to UCCX (you can see a demo video I put
>> together showcasing functionality, including export to Visio and AEF).
>> After much time I'm extremely pleased to announce we're looking for beta
>> testers to play with the software and provide feedback.
>>
>> Ideally I'm looking for folks who are able to apply real world needs to
>> the application to make recommendations for improvement and enhancements.
>>
>> Anyone interested please feel free to email me directly or on thread to
>> discuss further.
>>
>> The software is Web based, so if you're a Mac or Linux guy, you can still
>> apply!
>>
>> I'm also happy to answer any questions out there regarding the software.
>>
>> Regards,
>> Tanner Ezell
>>
>
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Re: [cisco-voip] Cisco Voice Training

[Pavan K]

Vik malhi has an excellent course.
Faisal Khan is iffy at best.
On May 3, 2016 1:50 PM, "Brian Meade"  wrote:

> The CBTNuggets CCNP Voice/Collab videos are really good at a lot of this
> type of material.
>
> On Tue, May 3, 2016 at 2:43 PM, Aaron Jenkins 
> wrote:
>
>> Certification, we dropped yearly third party support/Maintenance on our
>> VoIP and I am only experienced in some parts of CUCM. Never setup up
>> gateways or Call routing and such.
>>
>>
>>
>> *From:* Ed Leatherman [mailto:ealeather...@gmail.com]
>> *Sent:* Tuesday, May 03, 2016 2:28 PM
>> *To:* Aaron Jenkins
>> *Cc:* cisco-voip@puck.nether.net
>> *Subject:* Re: [cisco-voip] Cisco Voice Training
>>
>>
>>
>> Are you after a cert or looking for some particular skill set?
>>
>>
>>
>> I'm using videos on ine.com to study for a re-certification, their
>> collab videos seem pretty solid but they are not cisco "authorized" so i
>> can't comment on how much they toe the line.
>>
>>
>>
>> On Mon, May 2, 2016 at 4:04 PM, Aaron Jenkins 
>> wrote:
>>
>> Looking for recommendations on Training Facilities to take Cisco Voice
>> classes.
>>
>>
>>
>> Thanks.
>>
>>
>>
>>
>>
>>
>>
>>
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>>
>>
>>
>>
>>
>> --
>>
>> Ed Leatherman
>>
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>>
>>
>
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Re: [cisco-voip] CUCM AD Integration

[Pavan K]

You can import users from 2 different domains but you can only authenticate
against one domain. If you need to authenticate across multiple domains
then you need an LDS / ADAM instance

On Wed, Jun 3, 2015, 08:16 Michel L. M. B. Perez michelmbpe...@gmail.com
wrote:

 Hi guys,

 I have a question, i think this is not so usual, not for me.

 If i have my CUCM enviroment, like a cluster on version 9.1 with 1 PUB and
 3 Subscribers.

 My customer is using a domain with @contoso.com this CUCM cluster is
 importing LDAP users from doamin @contoso.com. My question is can i map
 another domain, completely different with no relationship like @
 whereis.com to import this users to my CUCM cluster and this users can
 use my CUCM / Unity / Jabber structure?

 Or will i need to create another enviroment for this other domain?

 Thanks.
 --
 Michel Perez
 Skype: michelmbperez
 michelmbpe...@gmail.com
 http://br.linkedin.com/in/michelmbperez
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Re: [cisco-voip] Hunt Group?

[Pavan K]

if you have them in the same partition then you have a shared line and they
will ring at once.
If you have them in different partition, you can have them ring one after
another depending on the call distribution algorithm you want to follow but
you lose the shared line functionality.

If you need both, you could do option 2 with Call pickup so that way phone
2 gets some kind of an audible notification on phone 2( of the call on
phone 1) but it can get very complicated pretty quickly.



On Wed, Jan 7, 2015 at 6:18 PM, Haas, Neal nh...@co.fresno.ca.us wrote:

  The easiest thing to do is to have 4701 ring, then role to 4702, the
 role to 4703 etc.



 Then change the phone mask on all to read “4701”. Easiest, quickest, fix
 for the situation.





 Neal Haas



 *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On Behalf
 Of *Anthony Holloway
 *Sent:* Wednesday, January 07, 2015 12:17 PM
 *To:* Lisa Notarianni; cisco-voip@puck.nether.net
 *Subject:* Re: [cisco-voip] Hunt Group?



 Not if both DNS are in the same partition. But if you change that, then
 you no longer have a shared line.

 On Wed, Jan 7, 2015 at 2:13 PM Lisa Notarianni 
 lisa.notaria...@scranton.edu wrote:

 Currently at Call manager version 8.6:



 Does anyone know of a way to have an extension ring on one phone 3 times
 or so and then have the same extension ring on another phone if no one
 answers?



 Ex:

 Phone A has DN 4701 and wants it to ring there 3 times



 then –



 On Phone B have it ring there on the 4th ring on DN 4701 if they did not
 answer on Phone A



 I have set Hunt Groups up but not using the same DN.  Not sure this can be
 done.



 Thanks.





 [image: LNsignatureFile]



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-- 
- Pavan
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Re: [cisco-voip] Jabber persistent chat history

[Pavan K]

Interesting.
Chat history is maintained locally on your machine. Are you saying that
your chat history is persisted across devices ?
We haven't experienced that on our deployment.

Pavan
On Nov 19, 2014 8:06 AM, Josh Warcop j...@warcop.com wrote:

  Persistent chat isn't the same as chat rooms. Persistency is maintained
 across clients via the Postgre Sql database that is attached to the IMP
 server.

 I believe there are some client type dependencies, but every time I look
 at my Jabber client and open up an IM conversation my chat history is still
 there. Regardless of which machine I use.

 Sent from my Windows Phone
  --
 From: Matthew Collins mcoll...@block.co.uk
 Sent: ‎11/‎19/‎2014 5:00 AM
 To: Josh Warcop j...@warcop.com; Pavan K pav.c...@gmail.com;
 cisco-voip@puck.nether.net
 Subject: RE: [cisco-voip] Jabber persistent chat history

   That’s not very scalable though as a room would need to be set up prior
 and then for every separate user you wanted this for or have I got that
 wrong.



 I know there was a feature on Lync called IM forking that would provide
 that exact feature. I did look into this a while ago and Cisco do provide
 IM Forking but only until the user reply’s.



 *IM Forking*

 When a user sends an IM to a contact who is signed in to multiple IM
 clients. IM and Presence Service delivers the IM to each client. This
 functionality is called IM forking. IM and Presence Service continues to
 fork IMs to each client, until the contact replies. Once the contact
 replies, IM and Presence Service only delivers IMs to the client on which
 the contact replied.

 Matt



 *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On Behalf
 Of *Josh Warcop
 *Sent:* 19 November 2014 01:07
 *To:* Pavan K; cisco-voip@puck.nether.net
 *Subject:* Re: [cisco-voip] Jabber persistent chat history



 That is exactly what persistent chat provides.

 Sent from my Windows Phone
   --

 *From: *Pavan K pav.c...@gmail.com
 *Sent: *‎11/‎18/‎2014 6:49 PM
 *To: *cisco-voip@puck.nether.net
 *Subject: *[cisco-voip] Jabber persistent chat history

 Is there a way to do persistent chat across jabber clients

 So let's say I am in a jabber conversation on my laptop, can I resume the
 conversation on my jabber mobile and have my conversation history from my
 laptop follow me to my mobile.

 Basically like a persistent group chat without a chat room.

 Pavan

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[cisco-voip] Jabber contact disappears consistently

[Pavan K]

We have an interesting problem on a new jabber deployment that has us
stumped. Wonder if anybody else saw this.

Two jabber servers with ha enabled and balanced users. Using 10.5su1 for
jabber windows and ucm/imp.

Leveraging sip directory uri as the IM scheme due to a multi forest
environment with duplicate Samaccountnames across domains.

UserA contact list has userB on it. Folks can im each other without any
problem. If we move userB from his imp server to another server in the same
subcluster, userB disappears from userA's contact list.

Repeatable across multiple users with different userA and userB and happens
every time regardless of moving them from server1 to server2 or vice versa.

Using router to router communication between nodes and default
jabber-config.

Any ideas ?
Have a TAC case open but its going nowhere.
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[cisco-voip] Jabber persistent chat history

[Pavan K]

Is there a way to do persistent chat across jabber clients

So let's say I am in a jabber conversation on my laptop, can I resume the
conversation on my jabber mobile and have my conversation history from my
laptop follow me to my mobile.

Basically like a persistent group chat without a chat room.

Pavan
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[cisco-voip] TAPS with E164

[Pavan K]

I am trying to deploy a new site on UCM 9.1 containing about 1000 users The
directory numbers for the phones are full E164 including the leading +. I
was planning on deploying UCCX for TAPS.

Would users be able to enter the e164 DN through TAPS IVR ?
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Re: [cisco-voip] Phone question

[Pavan K]

Set up the ongoing call on a meet me bridge.

Have a speed dial/plar from the phone to the meet me number.

Pavan
 On Oct 21, 2014 2:51 PM, Haas, Neal nh...@co.fresno.ca.us wrote:

  Is there a way to have a phone when picked up automatically conferenced
 into an ongoing call?



 We have a walk up counter they call a lot into a translator service, we
 want to make it easy for the client to get onto the call. Simply just pick
 up the handset.



 Is this possible?





 Neal Haas



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Re: [cisco-voip] Log Files

[Pavan K]

Grep through all files for AlarmErr
On Aug 6, 2014 12:17 PM, Carlo via cisco-voip cisco-voip@puck.nether.net
wrote:

 What I was looking for is which of the 30 or so files has the Media
 resource logs in it.

 Sent from my iPad

 On Aug 6, 2014, at 7:52 AM, Brian Meade bmead...@vt.edu wrote:

 Set CallManager traces to Detailed on all nodes then pull Cisco
 CallManager Traces using RTMT under TraceLog Central-Collect Files next
 time the issue occurs.


 On Wed, Aug 6, 2014 at 10:41 AM, Carlo via cisco-voip 
 cisco-voip@puck.nether.net wrote:

 I have CUCM 8.6 and getting media resources exhausted.

 Which log file can I look at to see were the problem is.

 Thanks


 Carlo

 Sent from my iPad
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Re: [cisco-voip] UCCE Reporting

[Pavan K]

Read the db schema and the reporting guide.
Nothing can replace real world experience, but that should get you at least
half the way.
On Jul 11, 2014 11:27 AM, Nortel Nan via cisco-voip 
cisco-voip@puck.nether.net wrote:

 I'm an independent contractor and I'm diving into the UCCE world.  I just
 finished the formal training class but it didn't include the reporting
 element of the product.  I'm very good at reading technical manuals and
 applying the instructions.  So my question is:  how hard is the reporting
 to learn?  Where are the best links to the technical information?  Are
 there other web links that you've found useful?

  Thanks,
 Nortel Nan

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Re: [cisco-voip] what do i lose with the publisher being down?

[Pavan K]

You don't lose much.

I believe any write operations on the db will fail.
Example - trying to set CFA from phone.
On Jul 11, 2014 4:49 PM, Ted Nugent tednugen...@gmail.com wrote:

 Also I've run into a problem with EM where users attempted to logout from
 sub service URL and could not however that may have been a bug with version
 we were on so it may be a good idea to test that if that's a concern.


 On Fri, Jul 11, 2014 at 3:50 PM, Brian Meade bmead...@vt.edu wrote:

 With internal services provisioning, phones now reach out to their
 subscriber for things such as Directories.

 Make sure the publisher isn't the only MTP/MOH/CFB resource for any MRGLs.

 Make sure the gateways are set to send calls to the subscriber primarily
 if using H.323/SIP.


 On Fri, Jul 11, 2014 at 2:27 PM, Lelio Fulgenzi le...@uoguelph.ca
 wrote:


 I'm hoping I can get some real world comments here...

 I need to take the publisher down in order to move it. From what I
 understand, for the most part, service will continue to operate. However,
 some things will not be available. From what I gather:

- CUCMUser pages will not be available
- CUCMAdmin pages will not be available
- Services Subscription listing when Services URL pointing to the
publisher

 What else will be unavailable? We are not using extension mobility.


 If I program the service URL of a phone to point to a different server,
 will the phone continue to call that separate server? Or does it need the
 publisher to get that information first?


 Thoughts?

 ---
 Lelio Fulgenzi, B.A.
 Senior Analyst, Network Infrastructure
 Computing and Communications Services (CCS)
 University of Guelph

 519‐824‐4120 Ext 56354
 le...@uoguelph.ca
 www.uoguelph.ca/ccs
 Room 037, Animal Science and Nutrition Building
 Guelph, Ontario, N1G 2W1


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Re: [cisco-voip] International Calls

[Pavan K]

You can get a sip trunk from a low cost provider and route Intl calls
there. Lot less hassle that way.
On Jul 8, 2014 10:00 AM, Jose Colon II jcolon...@gmail.com wrote:

 Need some advice for international calls.

 I want to be able to make international calls as easy as picking a line
 off the phone but using a service like vonage to complete the call. I want
 to be able to allow multiple people to use this line from different parts
 of the building so it would look like below.

 Example. Hook the Vonage device to a FXO card that then can be used from a
 cisco phone by picking line 2 on a cisco phone.

 The service does not have to be vonage. I am just looking for a way to
 make international calls cheaper and to allow multiple people to use it.

 Anyone doing anything like this?



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Re: [cisco-voip] Call Manager Active/Inactive version

[Pavan K]

Jose,
That is not supported. If you are using a VM, you could always export the
virtual machine out.
If you are on physical hardware, some old schoolers create backups by
pulling out one of the disks in the raid array as an offline backup.
 On Jul 8, 2014 4:57 PM, Jose Colon II jcolon...@gmail.com wrote:

 Is there a way to install call manager into the inactive partition?

 Here is my thinking. Lets say call manager on the active partition goes
 down and is not recoverable. It would be beneficial to switch to the
 inactive partition and restore from backup. This obviously has some
 benefits as a standby system ready to be turned on.

 Thanks

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Re: [cisco-voip] dspfarm profile 1 conference video homogenous

[Pavan K]

Jason,
I don't think IBS was implemented in CME. it was implemented in UCM 7.0 if
I am not mistaken.

Pavan
On Jun 6, 2014 5:00 PM, Jason Aarons (AM) jason.aar...@dimensiondata.com
wrote:

  Brian, does this CUCM “intelligent bridge selection” hold true for
 CallManager Express 9.1?  CME doesn’t appear intelligent enough to even
 consider this profile for usage!  See debug ephone hw-conference below.



 I setup a dspfarm profile 1 conf video homo and its active/registered to
 CME 9.1, however my phones can’t do any conferencing (cannot complete
 conference).  Right now this is audio only.  I have a PVDM3-256 in here and
 a 10 channel PRI.



 Debug sccp all I don’t even see the CME phones trying to use it;



 IOS 15.2.5M



 voice-card 0

 voice-service dsp-reservation 40

 dspfarm

 dsp services dspfarm

 !

 dspfarm profile 1 conference video homogeneous

 codec g729br8

 codec g729r8

 codec g729abr8

 codec g729ar8

 codec g711alaw

 codec g711ulaw

 codec ilbc

 codec h264 vga frame-rate 30 bitrate 1mbps

 maximum sessions 8

 associate application SCCP

 !

 Show sccp

 2911-CDG#show sccp

 SCCP Admin State: UP

 Gateway Local Interface: Vlan502

 IPv4 Address: 10.82.66.254

 Port Number: 2000

 IP Precedence: 5

 User Masked Codec list: None

 Call Manager: 10.82.66.254, Port Number: 2000

 Priority: 1, Version: 7.0, Identifier: 1



 V_Conferencing Oper State: ACTIVE - Cause Code: NONE

 Active Call Manager: 10.82.66.254, Port Number: 2000

 TCP Link Status: CONNECTED, Profile Identifier: 1

 Reported Max Streams: 32, Reported Max OOS Streams: 0

 Layout: default 1x1

 Supported Codec: g729br8, Maximum Packetization Period: 60

 Supported Codec: g729r8, Maximum Packetization Period: 60

 Supported Codec: g729abr8, Maximum Packetization Period: 60

 Supported Codec: g729ar8, Maximum Packetization Period: 60

 Supported Codec: g711alaw, Maximum Packetization Period: 30

 Supported Codec: g711ulaw, Maximum Packetization Period: 30

 Supported Codec: ilbc, Maximum Packetization Period: 120

 Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30

 Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30

 Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization
 Period: 30

 Supported Codec: h264: 2.0 (vga), Frame Rate: 30fps, Bit Rate: 1000-1000
 Kbps

Max MBPS: 72 (x500 MB/s) Max FS: 5 (x256 MBs)

 TLS : ENABLED



 show run | begin ephone-dn  7

 ephone-dn  7  octo-line

 number 9AAA

 conference ad-hoc video

 !



 show ephone-dn conference ad-hoc

 type active inactive  numbers

 ===

 Ad-hoc Video0   8  9AAA

 DN tags: 7





 Here is a debug ephone hw-conference

 Jun  6 21:49:20.674: SkinnyHWConfAPI: reqType 17:Is XFR to Conf

 Jun  6 21:49:20.678: SkinnyHWConfAPI: reqType 12:Hold On

 Jun  6 21:49:20.678: skinny_hwconf_video_hold:Hold dn:ch 1:1

 Jun  6 21:49:20.678: SkinnyHWConfAPI: reqType 48:Video DN type

 Jun  6 21:49:20.682: SkinnyHWConfAPI: reqType 24:Codec Switch

 Jun  6 21:49:23.486: SkinnyHWConfAPI: reqType 33:Meetme Opened

 Jun  6 21:49:23.862: SkinnyHWConfAPI: reqType 33:Meetme Opened

 Jun  6 21:49:23.994: SkinnyHWConfAPI: reqType 33:Meetme Opened

 Jun  6 21:49:24.306: SkinnyHWConfAPI: reqType 46:MTP for CAP list

 Jun  6 21:49:24.306: skinny_hwconf_preselect_mtp_tag_for_video:DN not
 video  cdn 1 cchan 2

 Jun  6 21:49:24.306: SkinnyHWConfAPI: reqType 17:Is XFR to Conf

 Jun  6 21:49:24.306: SkinnyHideAdhocConfNumber for dn=1 chan=2 streamID=0

 Jun  6 21:49:27.182: SkinnyHWConfAPI: reqType 17:Is XFR to Conf

 Jun  6 21:49:27.494: SkinnyHWConfAPI: reqType 23:Codec List

 Jun  6 21:49:27.498: Check associated hwconf with callid 3244

 Jun  6 21:49:27.498: Unable to associate a dn/chan with callid 3244

 Jun  6 21:49:27.498: Check associated hwconf with callid 3243

 Jun  6 21:49:27.498: SkinnyHWConfAPI: reqType 24:Codec Switch

 Jun  6 21:49:27.498: SkinnyHWConfAPI: reqType 50:Check to Open MM

 Jun  6 21:49:27.582: SkinnyHWConfAPI: reqType 19:Bridge Media

 Jun  6 21:49:27.582: SkinnyHwconfBridgeMedia:dn = 1, chan 2, phone = 0
 stream 0

 Jun  6 21:49:27.582: SkinnyHwconfBridgeMedia:Call not in conference

 Jun  6 21:49:28.182: SkinnyHWConfAPI: reqType 17:Is XFR to Conf

 Jun  6 21:49:28.182: SkinnyHideAdhocConfNumber for dn=1 chan=2 streamID=0

 Jun  6 21:49:28.182: SkinnyHWConfAPI: reqType 17:Is XFR to Conf

 Jun  6 21:49:29.182: SkinnyHWConfAPI: reqType 17:Is XFR to Conf

 Jun  6 21:49:29.818: SkinnyAddDnToHwConference: phone=0 (dn=1 chan=1
 state=9) (conf_dn=1 conf_chan=2 state=4)

 Jun  6 21:49:29.818: SkinnyHWConfAPI: reqType 14:Is Conf

 Jun  6 21:49:29.818: SkinnyHWConfAPI: reqType 14:Is Conf

 Jun  6 21:49:29.818: SkinnyAddDnToHwConference: New adhoc conference

 Jun  6 21:49:29.822: SkinnyHWConfAPI: reqType 1:New Conf

 Jun  6 21:49:29.822: skinny_hwconf_new_adhoc_setup_video_call: Audio
 Ad-Hoc call

 Jun  6 

Re: [cisco-voip] DN report including CSS

[Pavan K]

The numplan table has the info.
Dn is under the dnorpattern field.

Run SQL select dnorpattern, any-other- field from numplan

On May 6, 2014 10:23 AM, Chris Chamberlain chamb...@oakland.edu wrote:

 Does anyone know if there is a way in CUCM 9.1 to generate a report
 detailing a list of DNs and their assigned CSS?  This seems like it would
 be something fairly straightforward, however we are not seeing a way to do
 this.  We have been asked to find every DN that currently has International
 Calling capability, but without looking at each DN individually I do not
 see an automated way of obtaining this information.  Any Thoughts?

 -Chris

 Chris Chamberlain
 Associate Director Network Engineering
 Oakland University
 248.370.4321




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Re: [cisco-voip] Cisco 6961

[Pavan K]

Look at the UCM logs. It should have a registration rejected cause.

Could be a replication issue, device type mismatch, an itl issue or
something else. The UCM logs will point you in the right direction.
On Apr 11, 2014 9:03 AM, costas georgiou ckos1...@hotmail.com wrote:

 HI All,

 I am having issues trying to register a Ciso 961 IP Phone, it comes up as
 rejected.  I have entered the phone in CUCM.  The phone is picking up a IP
 address, the switch configuration is correct.  We have even movedit to a
 working port. Below is the show CDP output:

 Device ID: SEP44ADD9BD1D7D
 Entry address(es):
   IP address: 2.x.x.x
 Platform: Cisco IP Phone 6961 ,  Capabilities: Host Phone Two-port Mac
 Relay
 Interface: FastEthernet0/14,  Port ID (outgoing port): Port 1
 Holdtime : 143 sec
 Second Port Status: Down
 Version :
 SCCP 9.4.1.3
 advertisement version: 2
 Duplex: full
 Power drawn: 6.000 Watts
 Power request id: 10524, Power management id: 3
 Power request levels are:6000 0 0 0 0
 Management address(es):

 I have noticed 2nd port status is down, not sure if thi has anything to do
 with it.  We are having a problem at the momet with 6961 where the time is
 out by 1 hour and they are all running the sam firmware SCCP 9.4.1.3.  I am
 a bit stumped now, any ideas?

 Regards

 Costas


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Re: [cisco-voip] Delayed ringing

[Pavan K]

I would create a new ringtone which is silent for the first two secs and
set it on the phone.
On Apr 6, 2014 12:32 PM, Nortel Nan nortel...@aol.com wrote:

 I need an extension to appear on a 7962 but not audibly ring for the first
 two ring cycles.  How can I accomplish that?

  Nortel Nan
 nortel...@aol.com

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Re: [cisco-voip] Recording Proiles

[Pavan K]

Leslie,
When the speed dial is depressed, which line is being used to make the call
?
Perhaps that line does not have a. Recording profile. What happens when you
dial the same number manually on the same line ?

I am reasonably sure that speed dials do not have anything to do with
recording profiles.
On Mar 27, 2014 7:18 PM, Erick erick...@gmail.com wrote:

 Is there only one line (dn) on the phone or multiple? If multiple maybe
 the speed dial is using a line without a recording profile.

 I don't have a 88xx handy but will try from a speed dial on a 99xx if I
 get a chance tomorrow.

 Sent from my iPhone

 On Mar 27, 2014, at 5:33 PM, Leslie Meade leslie.me...@lvs1.com wrote:

  I have a client that is using the 88XX model of phone with KEM's. They
 also have a recording profile configured and working with no issues.

 However when there is a speed dial configured(either on the device or the
 KEM) the call is not being recorded. When we place a packet capture on the
 Nice software we do not see and packets hitting it.

 Use a line and we see traffic being sent to the server. Now is it due to
 we can not a recording profile on a speed dial that is causing this ? or
 the BIB is not firing up when the speed dial is selected. I have not
 checked on the server as of yet.





 If we span the recording the call gets recorded.



 Leslie



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Re: [cisco-voip] Recording Proiles

[Pavan K]

Leslie,
If you pull the call manager traces and pull out the Recording( instance
for the failed call.

Do you see any bib allocation error or any other errors from the recording
feature ?
On Mar 27, 2014 7:31 PM, Leslie Meade leslie.me...@lvs1.com wrote:

  Every line has a recording profile on it. If I select the speed dial it
 will select the primary line.

 If I make a call from the primary line it records that call. It's only
 speed dial that's not functional.







 *Leslie Meade*

 ..
 * Mobile:778.228.4339 778.228.4339* | *Main:* *604.676.5239
 604.676.5239*
 *Email:* leslie.me...@lvs1.com



 *From:* Pavan K [mailto:pav.c...@gmail.com]
 *Sent:* Thursday, March 27, 2014 5:29 PM
 *To:* Erick
 *Cc:* Leslie Meade; cisco-voip (cisco-voip@puck.nether.net)
 *Subject:* Re: [cisco-voip] Recording Proiles



 Leslie,
 When the speed dial is depressed, which line is being used to make the
 call ?
 Perhaps that line does not have a. Recording profile. What happens when
 you dial the same number manually on the same line ?

 I am reasonably sure that speed dials do not have anything to do with
 recording profiles.

 On Mar 27, 2014 7:18 PM, Erick erick...@gmail.com wrote:

 Is there only one line (dn) on the phone or multiple? If multiple maybe
 the speed dial is using a line without a recording profile.



 I don't have a 88xx handy but will try from a speed dial on a 99xx if I
 get a chance tomorrow.


 Sent from my iPhone


 On Mar 27, 2014, at 5:33 PM, Leslie Meade leslie.me...@lvs1.com wrote:

  I have a client that is using the 88XX model of phone with KEM's. They
 also have a recording profile configured and working with no issues.

 However when there is a speed dial configured(either on the device or the
 KEM) the call is not being recorded. When we place a packet capture on the
 Nice software we do not see and packets hitting it.

 Use a line and we see traffic being sent to the server. Now is it due to
 we can not a recording profile on a speed dial that is causing this ? or
 the BIB is not firing up when the speed dial is selected. I have not
 checked on the server as of yet.





 If we span the recording the call gets recorded.



 Leslie



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Re: [cisco-voip] UCM 9.1 SIP Normalization Rules

[Pavan K]

Matt,

Here is a script i wrote to get this done a long time back.
This was to strip the 5060 from the request URI going towards an IMS core
on an ISC trunk but it should work for you.


==


-- Strip off the port information from the Request URI


local function stripPortFromRequestURI(msg)




-- Get the URI from the SIP Request


local method, uri, ver = msg:getRequestLine()




if uri


then


-- Split the uri into SIP URI  URI Parameters


local newuri, params = string.match(uri, ([^;]*)(;.*))




if not newuri


then


newuri = uri


end




-- Get the USER  the HOST part of the URI


local lhs, rhs = string.match(newuri, (.*)@(.*))




local _, numOfColons = string.gsub(rhs, :, :)




-- If there are minimum of 2 :, then the host part is an IPv6
Address


if numOfColons = 2


then


-- This is a n IPv6 Address




-- URI with IPv6 address will be of the form -
sip:user:password@[1234::5678]:5060


-- Check if the host part contains ], if so, port number
might be present


rhs = string.gsub(rhs, (%[.*%]):.*, %1)


else


-- This is an IPv4 address or a FQDN




-- Strip off the characters after : (i.e) the port number


rhs = string.gsub(rhs, (.*):.*, %1)


end




-- Generate the modified URI


newuri = string.format(%s@%s%s, lhs, rhs, params or )




-- Set the new SIP URI


msg:setRequestUri(newuri)


end


end




M.outbound_200_INVITE  = stripPortFromRequestURI


M.outbound_INVITE  = stripPortFromRequestURI




return M





On Fri, Mar 14, 2014 at 8:37 AM, Matt Slaga (AM) 
matt.sl...@dimensiondata.com wrote:

 The port number is actually something entirely different.  In this case, I
 have 70 SIP trunks between UCM and Lync for a global egress point access
 (CSS different for each SIP trunk).  The ports are actually in the
 5020-5050 range.



 UCM puts this request in the SIP Start Line (Invite) with 5060 regardless
 of what the SIP trunk uses.



 Everything works fine if the user is part of the Lync pool that is
 associated with the Mediation server.  If they are on a different pool,
 this port of 5060 (not talking transport here, just SIP message adjustment)
 causes Lync to drop the invite.  When this port reads 5061 in the SIP
 invite, the calls flow properly.







 *From:* Florian Kroessbacher [mailto:florian.kroessbac...@gmail.com]
 *Sent:* Friday, March 14, 2014 9:24 AM
 *To:* Matt Slaga (AM)
 *Cc:* 'Cisco-Voip-Puck' (cisco-voip@puck.nether.net)
 *Subject:* Re: [cisco-voip] UCM 9.1 SIP Normalization Rules





 what about to change the destination port in the siptrunk config ??

 --
 Florian Kroessbacher
 gmail: florian.kroessbac...@gmail.com



 On Fri, Mar 14, 2014 at 2:20 PM, Matt Slaga (AM) 
 matt.sl...@dimensiondata.com wrote:

 I'm attempting to apply a SIP Normalization rule to make a port adjustment
 on SIP calls between UCM and Lync 2013.



 In the header, UCM sends the port number as 5060.  Lync has problems
 digesting this when referring the call to another pool and wants to see the
 header at 5061.



 The script we are attempting to use is below.  It has been applied to the
 various SIP trunks between UCM and Lync.



 When reviewing traces, the port is not being adjusted.  Either the LUA
 script is wrong below, or for some reason it is not being applied.



 Any thoughts?





 M = {}
 function M.outbound_INVITE(msg)
 local method, ruri, ver = msg:getRequestLine()
 local uri = string.gsub(ruri, 5060, 5061)
 msg:setRequestUri(uri)
 end
 return M





 image001.png



 image002.png







 Matt Slaga

 Dimension Data

 Tel:+1-571-203-4132

 matt.sl...@dimensiondata.com






 itevomcid

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-- 
- Pavan
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Re: [cisco-voip] Setting up ICM Sprawler 8.x

[Pavan K]

Ryan,
If you are going to put it up on EC2, you should be able to put the ad on
the same box.

I  havent used EC2 yet for uc so am curious, wouldn't it be cheaper to
build a server ?
On Mar 6, 2014 12:54 PM, Stephen Welsh stephen.we...@unifiedfx.com
wrote:

  I plan to setup a Sprawler at some point too, the following looks like a
 useful guide:

  https://supportforums.cisco.com/docs/DOC-1374

  I states the AD/DNS can be on the same box, however the post is a few
 years old, is this something that changes with a particular release?

  It would be great if there was an updated version of the above guide as
 I know there are usually a ton of caveats and special settings that need
 to be taken into consideration.

  Cheers

  Stephen

  On 5 Mar 2014, at 19:12, Beck, Christopher cb...@usg.com wrote:

   Pavan is right.   You will need SQL because the router/logger processes
 are on the server.  If you install the AW/HDS service on a different host,
 that will require SQL as well.  You will need separate server(s) for the
 AD/DNS.  It isn't supported as a function installed on the IPCC servers.


  It doesn't matter if you have a sprawler or a multi-server production
 setup, these rules apply.

  Hope it helps,
  Chris

  *From:* cisco-voip 
 [mailto:cisco-voip-boun...@puck.nether.netcisco-voip-boun...@puck.nether.net
 ] *On Behalf Of *Ryan Burtch
 *Sent:* Wednesday, March 05, 2014 12:41 PM
 *To:* Pavan K
 *Cc:* cisco-voip@puck.nether.net
 *Subject:* Re: [cisco-voip] Setting up ICM Sprawler 8.x

  This is for a Sprawler.  Do you have any experience setting one of these
 up?




   Sincerely,

   Ryan Burtch


  On Wed, Mar 5, 2014 at 1:38 PM, Pavan K pav.c...@gmail.com wrote:

  AD /DNS is usually on a separate box
 You need SQL anywhere on the logger / aw/HDS/dds boxes. So yes you will
 need SQL.
   On Mar 5, 2014 1:03 AM, Ryan Burtch rburt...@gmail.com wrote:

   I have a couple questions on the ICM Sprawler setup.

   1. Does SQL need to be installed on the Sprawler?
   2. Can AD/DNS be installed on the Sprawler?




   Sincerely,

   Ryan Burtch


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Re: [cisco-voip] Setting up ICM Sprawler 8.x

[Pavan K]

AD /DNS is usually on a separate box
You need SQL anywhere on the logger / aw/HDS/dds boxes. So yes you will
need SQL.
On Mar 5, 2014 1:03 AM, Ryan Burtch rburt...@gmail.com wrote:

 I have a couple questions on the ICM Sprawler setup.

 1. Does SQL need to be installed on the Sprawler?
 2. Can AD/DNS be installed on the Sprawler?




 Sincerely,

 Ryan Burtch

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